@@ -68,6 +68,7 @@ OBJS-$(CONFIG_ANULL_FILTER) += af_anull.o
OBJS-$(CONFIG_APAD_FILTER) += af_apad.o
OBJS-$(CONFIG_APERMS_FILTER) += f_perms.o
OBJS-$(CONFIG_APHASER_FILTER) += af_aphaser.o generate_wave_table.o
+OBJS-$(CONFIG_APITCH_FILTER) += af_apitch.o
OBJS-$(CONFIG_APULSATOR_FILTER) += af_apulsator.o
OBJS-$(CONFIG_AREALTIME_FILTER) += f_realtime.o
OBJS-$(CONFIG_ARESAMPLE_FILTER) += af_aresample.o
new file mode 100644
@@ -0,0 +1,764 @@
+/*
+ * Copyright (c) 2019 Paul B Mahol
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU Lesser General Public License as published
+ * by the Free Software Foundation; either version 2.1 of the License,
+ * or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "libswresample/swresample.h"
+#include "libavutil/avassert.h"
+#include "libavutil/avstring.h"
+#include "libavutil/audio_fifo.h"
+#include "libavfilter/internal.h"
+#include "libavutil/common.h"
+#include "libavutil/opt.h"
+#include "libavcodec/avfft.h"
+#include "filters.h"
+#include "audio.h"
+
+typedef struct APitchContext {
+ const AVClass *class;
+
+ float pitch;
+ float tempo;
+ int window_size;
+ int ratio;
+
+ int power_change;
+ int fft_bits;
+ int nb_channels;
+
+ FFTContext *fft, *ifft;
+ AVAudioFifo *ififo;
+ int64_t pts;
+ int eof;
+ float *window_func_lut;
+
+ AVFrame *buffer;
+ AVFrame *magnitude;
+ AVFrame *phase;
+ AVFrame *acc;
+ AVFrame *new_phase;
+ AVFrame *last_phase;
+ AVFrame *osamples;
+ AVFrame *peaks;
+ AVFrame *map;
+
+ int input_overlap;
+ int output_overlap;
+ int samples_to_drain;
+ float last_power;
+
+ int flushed;
+ int pitch_changed;
+ struct SwrContext *swr;
+} APitchContext;
+
+#define OFFSET(x) offsetof(APitchContext, x)
+#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption apitch_options[] = {
+ { "pitch", "set pitch scale factor", OFFSET(pitch), AV_OPT_TYPE_FLOAT, {.dbl=1}, .01, 100, AF },
+ { "tempo", "set tempo scale factor", OFFSET(tempo), AV_OPT_TYPE_FLOAT, {.dbl=1}, .01, 100, AF },
+ { "oratio", "set overlap ratio", OFFSET(ratio), AV_OPT_TYPE_INT, {.i64=4}, 1, 64, AF },
+ { NULL },
+};
+
+AVFILTER_DEFINE_CLASS(apitch);
+
+static int set_input_overlap(APitchContext *s, float rate)
+{
+ s->input_overlap = s->output_overlap * rate;
+ if (s->input_overlap <= 0)
+ return AVERROR(EINVAL);
+ return 0;
+}
+
+static int config_output(AVFilterLink *outlink)
+{
+ AVFilterContext *ctx = outlink->src;
+ AVFilterLink *inlink = ctx->inputs[0];
+ APitchContext *s = ctx->priv;
+ int ret;
+
+ s->swr = swr_alloc();
+ if (!s->swr)
+ return AVERROR(ENOMEM);
+
+ s->swr = swr_alloc_set_opts(s->swr,
+ inlink->channel_layout, inlink->format, inlink->sample_rate,
+ inlink->channel_layout, inlink->format, inlink->sample_rate * s->pitch,
+ 0, ctx);
+ if (!s->swr)
+ return AVERROR(ENOMEM);
+
+ ret = swr_init(s->swr);
+ if (ret < 0)
+ return ret;
+
+ s->window_size = inlink->sample_rate / 10;
+ s->power_change = 1;
+ s->pts = AV_NOPTS_VALUE;
+ s->fft_bits = av_log2(s->window_size);
+ s->fft = av_fft_init(s->fft_bits, 0);
+ s->ifft = av_fft_init(s->fft_bits, 1);
+ if (!s->fft || !s->ifft)
+ return AVERROR(ENOMEM);
+
+ s->window_size = 1 << s->fft_bits;
+
+ s->ratio = FFMIN(1 << av_log2(s->ratio), s->window_size >> 1);
+ s->nb_channels = outlink->channels;
+
+ s->ififo = av_audio_fifo_alloc(outlink->format, outlink->channels, s->window_size);
+ if (!s->ififo)
+ return AVERROR(ENOMEM);
+
+ s->window_func_lut = av_realloc_f(s->window_func_lut, s->window_size,
+ sizeof(*s->window_func_lut));
+ if (!s->window_func_lut)
+ return AVERROR(ENOMEM);
+
+ for (int i = 0; i < s->window_size; i++) {
+ float t = (float)i / (float)(s->window_size - 1);
+ float h = 0.5 * (1.0 - cosf(2.0 * M_PI * t));
+ s->window_func_lut[i] = h;
+ }
+
+ s->output_overlap = s->window_size / s->ratio;
+ if (s->output_overlap <= 0)
+ return AVERROR(EINVAL);
+
+ return set_input_overlap(s, s->tempo / s->pitch);
+}
+
+static float get_power(const float *in, const int size)
+{
+ float sum = 0.f;
+
+ for (int n = 0; n < size; n++)
+ sum += in[n] * in[n];
+
+ return sqrtf(sum / size);
+}
+
+static void normalize(float *new, const float num, const int size)
+{
+ float den = get_power(new, size);
+ float f = den > 0.f ? num / den : 0.f;
+
+ for (int i = 0; i < size; i++)
+ new[i] *= f;
+}
+
+static void complex2polar(const FFTComplex *in,
+ float *magn, float *phase, const int size)
+{
+ for (int i = 0; i < size; i++) {
+ const float re = in[i + 1].re;
+ const float im = in[i + 1].im;
+
+ magn[i] = hypotf(re, im);
+ phase[i] = atan2f(im, re);
+ }
+}
+
+static void polar2complex(FFTComplex *out,
+ const float *magn, const float *phase,
+ const int size)
+{
+ for (int i = 0; i < size; i++) {
+ const float m = magn[i];
+ const float p = phase[i];
+
+ out[i + 1].re = cosf(p) * m;
+ out[i + 1].im = sinf(p) * m;
+ }
+}
+
+static void update_phase(const float *phase, float *last_phase,
+ float *new_phase, const int size)
+{
+ for (int i = 0; i < size; i++) {
+ new_phase[i] = last_phase[i];
+ last_phase[i] = phase[i];
+ }
+}
+
+static void restore_symmetry(FFTComplex *win, const int size)
+{
+ for (int i = 1; i < size / 2; i++) {
+ int pos = size - i;
+
+ win[pos].re = win[i].re;
+ win[pos].im = -win[i].im;
+ }
+}
+
+static float principal_angle(const float a)
+{
+ float b = fmodf(a + M_PI, 2 * M_PI);
+
+ if (b < 0.f)
+ b = 2 * M_PI + b;
+ b -= M_PI;
+
+ return b;
+}
+
+static void make_phase(const float *phase, const float *last_phase,
+ float *new_phase,
+ int io, int oo, int size)
+{
+ const float size2 = 2 * size;
+
+ for (int i = 0; i < size; i++) {
+ float h = 2 * M_PI * (i + 1.f) / size2;
+ float ii = phase[i] - last_phase[i] - io * h;
+ float j = h + principal_angle(ii) / io;
+
+ new_phase[i] += oo * j;
+ }
+}
+
+static int is_peak(const float *magnitude, const int pos, const int search)
+{
+ const float cmag = magnitude[pos];
+
+ for (int i = 0; i <= search; i++)
+ if (!(i > pos) && magnitude[pos - i] > cmag)
+ return 0;
+ for (int i = 0; i <= search; i++)
+ if (magnitude[pos + i] > cmag)
+ return 0;
+
+ return 1;
+}
+
+static int find_peaks(const float *magnitude, float *peaks, const int search, const int size)
+{
+ int count = 0;
+
+ for (int i = 0; i < size - search; i++) {
+ if (is_peak(magnitude, i, search))
+ peaks[count++] = i;
+ }
+
+ return count;
+}
+
+static void interp_phase(const float *map, const float *peaks, const int nb_peaks,
+ const float *phase, const float *last_phase,
+ float *new_phase, const float io, const float oo, const int size)
+{
+ const float size2 = 2 * size;
+
+ for (int i = 0; i < nb_peaks; i++) {
+ int peak = peaks[i];
+ int l = map[peak];
+ float m, n, o;
+
+ if (FFABS(peak - l) > FFMIN(1, av_log2(i)) - 3)
+ l = peak;
+ m = 2.f * M_PI * (peak + 1.f) / size2;
+ n = phase[peak] - last_phase[l] - io * m;
+ o = m + principal_angle(n) / io;
+
+ new_phase[peak] = new_phase[l] + oo * o;
+ }
+}
+
+static void get_new_phase(const float *map, const float *phase,
+ float *new_phase, const float I, const int size)
+{
+ for (int i = 0; i < size; i++) {
+ int f = map[i];
+
+ new_phase[i] = new_phase[f] + I * (phase[i] - phase[f]);
+ }
+}
+
+static int lowest_valley(const float *magnitude,
+ const int start, const int size)
+{
+ float lowest_magn = magnitude[start];
+ int lowest = start;
+
+ for (int i = start + 1; i < size; i++) {
+ if (magnitude[i] < lowest_magn) {
+ lowest = i;
+ lowest_magn = magnitude[i];
+ }
+ }
+
+ return lowest;
+}
+
+static void map_peaks(const float *peaks, const int nb_peaks,
+ const float *magnitude, float *map, const int size)
+{
+ int e = 0, f;
+
+ for (f = 0; f < nb_peaks - 1; f++) {
+ int peak = peaks[f];
+ int next_peak = peaks[f+1];
+
+ for (int j = lowest_valley(magnitude, peak, next_peak); e <= j; e++)
+ map[e] = peak;
+ }
+
+ for (; e < size; e++)
+ map[e] = peaks[f];
+}
+
+static void stretch(APitchContext *s, int nb_samples,
+ int window_size, float sample_rate,
+ const float *indata, float *outdata, int ch)
+{
+ FFTComplex *window_buffer = (FFTComplex *)s->buffer->extended_data[ch];
+ float *phase = (float *)s->phase->extended_data[ch];
+ float *magnitude = (float *)s->magnitude->extended_data[ch];
+ float *acc = (float *)s->acc->extended_data[ch];
+ float *new_phase = (float *)s->new_phase->extended_data[ch];
+ float *last_phase = (float *)s->last_phase->extended_data[ch];
+ float *osamples = (float *)s->osamples->extended_data[ch];
+ float *peaks = (float *)s->peaks->extended_data[ch];
+ float *map = (float *)s->map->extended_data[ch];
+ const int output_overlap = s->output_overlap;
+ const int input_overlap = s->input_overlap;
+ const int half_window_size = window_size / 2;
+ const float ratio = output_overlap / (float)input_overlap;
+ const float wscale = 1.f / window_size;
+ float power;
+
+ if (s->pitch == s->tempo && s->tempo == 1.f) {
+ for (int k = 0; k < window_size; k++)
+ osamples[k] = indata[k];
+
+ goto copy;
+ }
+
+ for (int k = 0; k < window_size; k++) {
+ const float window = s->window_func_lut[k];
+
+ window_buffer[k].re = indata[k] * window;
+ window_buffer[k].im = 0.f;
+ }
+
+ power = get_power(indata, window_size);
+ if (power > 2.f * s->last_power)
+ s->power_change = 1;
+
+ av_fft_permute(s->fft, window_buffer);
+ av_fft_calc(s->fft, window_buffer);
+
+ complex2polar(window_buffer, magnitude, phase, half_window_size);
+
+ if (s->power_change)
+ update_phase(phase, last_phase, new_phase, half_window_size);
+ s->power_change = 0;
+
+ if (ratio < 2.f) {
+ int nb_peaks = find_peaks(magnitude, peaks, 1, half_window_size);
+ float I = 2.f / 3.f + fminf(1.5, ratio) / 3.f;
+
+ av_assert0(nb_peaks > 0);
+ map_peaks(peaks, nb_peaks, magnitude, map, half_window_size);
+ interp_phase(map, peaks, nb_peaks, phase, last_phase, new_phase,
+ input_overlap, output_overlap, half_window_size);
+ get_new_phase(map, phase, new_phase, I, half_window_size);
+ } else {
+ make_phase(phase, last_phase, new_phase, input_overlap, output_overlap, half_window_size);
+ }
+
+ polar2complex(window_buffer, magnitude, new_phase, half_window_size);
+ restore_symmetry(window_buffer, window_size);
+
+ av_fft_permute(s->ifft, window_buffer);
+ av_fft_calc(s->ifft, window_buffer);
+
+ for (int k = 0; k < window_size; k++)
+ osamples[k] = window_buffer[k].re * wscale;
+
+ normalize(osamples, power, window_size);
+
+copy:
+ for (int k = 0; k < window_size; k++) {
+ const float window = s->window_func_lut[k];
+ osamples[k] *= window * .5f;
+ }
+
+ for (int k = 0; k < window_size; k++) {
+ acc[k] += osamples[k];
+ }
+
+ for (int k = 0; k < output_overlap; k++)
+ outdata[k] = acc[k];
+
+ memmove(acc, acc + output_overlap, (window_size * 2 - output_overlap) * sizeof(float));
+ FFSWAP(uint8_t *, s->last_phase->extended_data[ch], s->phase->extended_data[ch]);
+
+ s->last_power = power;
+}
+
+static int filter_frame(AVFilterLink *inlink)
+{
+ AVFilterContext *ctx = inlink->dst;
+ AVFilterLink *outlink = ctx->outputs[0];
+ APitchContext *s = ctx->priv;
+ AVFrame *out = NULL, *in = NULL, *new_out = NULL;
+ int ret, drain, available = 0;
+
+ if (s->pitch_changed) {
+ out = ff_get_audio_buffer(outlink, s->output_overlap);
+ if (!out) {
+ return AVERROR(ENOMEM);
+ }
+
+ ret = swr_convert_frame(s->swr, out, NULL);
+ if (ret < 0) {
+ av_frame_free(&out);
+ return ret;
+ }
+
+ if (out->nb_samples > 0) {
+ out->pts = s->pts;
+ s->pts += out->nb_samples;
+
+ return ff_filter_frame(outlink, out);
+ } else {
+ s->pitch_changed = 0;
+ av_frame_free(&out);
+ }
+
+ s->swr = swr_alloc_set_opts(s->swr,
+ inlink->channel_layout, inlink->format, inlink->sample_rate,
+ inlink->channel_layout, inlink->format, inlink->sample_rate * s->pitch,
+ 0, ctx);
+ if (!s->swr)
+ return AVERROR(ENOMEM);
+
+ ret = swr_init(s->swr);
+ if (ret < 0)
+ return ret;
+ }
+
+ in = ff_get_audio_buffer(outlink, s->window_size);
+ if (!in)
+ return AVERROR(ENOMEM);
+
+ ret = av_audio_fifo_peek(s->ififo, (void **)in->extended_data, s->window_size);
+ if (ret < 0) {
+ ret = AVERROR(ENOMEM);
+ goto end;
+ }
+ available = av_audio_fifo_size(s->ififo) > 0;
+
+ if (available > 0) {
+ out = ff_get_audio_buffer(outlink, s->output_overlap);
+ if (!out) {
+ ret = AVERROR(ENOMEM);
+ goto end;
+ }
+
+ for (int ch = 0; ch < inlink->channels; ch++) {
+ stretch(s, in->nb_samples,
+ s->window_size,
+ inlink->sample_rate,
+ (const float *)in->extended_data[ch],
+ (float *)out->extended_data[ch], ch);
+ }
+ }
+
+ if (s->pitch != 1.f) {
+ new_out = ff_get_audio_buffer(outlink, s->output_overlap);
+ if (!new_out) {
+ av_frame_free(&out);
+ ret = AVERROR(ENOMEM);
+ goto end;
+ }
+
+ if (out)
+ out->sample_rate *= s->pitch;
+ ret = swr_convert_frame(s->swr, new_out, out);
+ av_frame_free(&out);
+ if (ret < 0)
+ goto end;
+ out = new_out;
+ new_out = NULL;
+ }
+
+ if (!out) {
+ ret = 1;
+ goto end;
+ }
+
+ out->pts = s->pts;
+ s->pts += s->output_overlap;
+
+ s->flushed = out->nb_samples == 0;
+ if (s->flushed) {
+ ret = 1;
+ av_frame_free(&out);
+ goto end;
+ }
+
+ ret = ff_filter_frame(outlink, out);
+ if (ret < 0)
+ goto end;
+
+ s->samples_to_drain = s->input_overlap;
+ drain = FFMIN(s->samples_to_drain, av_audio_fifo_size(s->ififo));
+ av_audio_fifo_drain(s->ififo, drain);
+ s->samples_to_drain -= drain;
+
+end:
+ av_frame_free(&in);
+ av_frame_free(&new_out);
+ return ret;
+}
+
+static int activate(AVFilterContext *ctx)
+{
+ AVFilterLink *inlink = ctx->inputs[0];
+ AVFilterLink *outlink = ctx->outputs[0];
+ APitchContext *s = ctx->priv;
+ AVFrame *in = NULL;
+ int ret = 0, status;
+ int64_t pts;
+
+ if (!s->magnitude) {
+ s->magnitude = ff_get_audio_buffer(outlink, s->window_size / 2);
+ if (!s->magnitude)
+ return AVERROR(ENOMEM);
+ }
+
+ if (!s->phase) {
+ s->phase = ff_get_audio_buffer(outlink, s->window_size / 2);
+ if (!s->phase)
+ return AVERROR(ENOMEM);
+ }
+
+ if (!s->acc) {
+ s->acc = ff_get_audio_buffer(outlink, s->window_size * 2);
+ if (!s->acc)
+ return AVERROR(ENOMEM);
+ }
+
+ if (!s->new_phase) {
+ s->new_phase = ff_get_audio_buffer(outlink, s->window_size / 2);
+ if (!s->new_phase)
+ return AVERROR(ENOMEM);
+ }
+
+ if (!s->last_phase) {
+ s->last_phase = ff_get_audio_buffer(outlink, s->window_size / 2);
+ if (!s->last_phase)
+ return AVERROR(ENOMEM);
+ }
+
+ if (!s->osamples) {
+ s->osamples = ff_get_audio_buffer(outlink, s->window_size);
+ if (!s->osamples)
+ return AVERROR(ENOMEM);
+ }
+
+ if (!s->peaks) {
+ s->peaks = ff_get_audio_buffer(outlink, s->window_size / 2);
+ if (!s->peaks)
+ return AVERROR(ENOMEM);
+ }
+
+ if (!s->map) {
+ s->map = ff_get_audio_buffer(outlink, s->window_size / 2);
+ if (!s->map)
+ return AVERROR(ENOMEM);
+ }
+
+ if (!s->buffer) {
+ s->buffer = ff_get_audio_buffer(outlink, s->window_size * 2);
+ if (!s->buffer)
+ return AVERROR(ENOMEM);
+ }
+
+ FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
+
+ if (!s->eof && av_audio_fifo_size(s->ififo) < s->window_size) {
+ ret = ff_inlink_consume_frame(inlink, &in);
+ if (ret < 0)
+ return ret;
+
+ if (ret > 0) {
+ ret = av_audio_fifo_write(s->ififo, (void **)in->extended_data,
+ in->nb_samples);
+ if (ret >= 0 && s->pts == AV_NOPTS_VALUE)
+ s->pts = in->pts;
+
+ if (s->samples_to_drain > 0) {
+ int drain = FFMIN(s->samples_to_drain, av_audio_fifo_size(s->ififo));
+ av_audio_fifo_drain(s->ififo, drain);
+ s->samples_to_drain -= drain;
+ }
+
+ av_frame_free(&in);
+ if (ret < 0)
+ return ret;
+ }
+ }
+
+ if ((av_audio_fifo_size(s->ififo) >= s->window_size) ||
+ ((av_audio_fifo_size(s->ififo) > 0 || !s->flushed) && s->eof)) {
+ ret = filter_frame(inlink);
+ if (av_audio_fifo_size(s->ififo) >= s->window_size)
+ ff_filter_set_ready(ctx, 100);
+ if (ret != 1)
+ return ret;
+ }
+
+ if (!s->eof && ff_inlink_acknowledge_status(inlink, &status, &pts)) {
+ if (status == AVERROR_EOF) {
+ s->eof = 1;
+ if (av_audio_fifo_size(s->ififo) >= 0 && !s->flushed) {
+ ff_filter_set_ready(ctx, 100);
+ return 0;
+ }
+ }
+ }
+
+ if (s->eof && (av_audio_fifo_size(s->ififo) <= 0 || s->flushed)) {
+ ff_outlink_set_status(outlink, AVERROR_EOF, s->pts);
+ return 0;
+ }
+
+ if (!s->eof)
+ FF_FILTER_FORWARD_WANTED(outlink, inlink);
+
+ return FFERROR_NOT_READY;
+}
+
+static int query_formats(AVFilterContext *ctx)
+{
+ AVFilterFormats *formats;
+ AVFilterChannelLayouts *layouts;
+ static const enum AVSampleFormat sample_fmts[] = {
+ AV_SAMPLE_FMT_FLTP,
+ AV_SAMPLE_FMT_NONE
+ };
+ int ret;
+
+ layouts = ff_all_channel_counts();
+ if (!layouts)
+ return AVERROR(ENOMEM);
+ ret = ff_set_common_channel_layouts(ctx, layouts);
+ if (ret < 0)
+ return ret;
+
+ formats = ff_make_format_list(sample_fmts);
+ if (!formats)
+ return AVERROR(ENOMEM);
+ ret = ff_set_common_formats(ctx, formats);
+ if (ret < 0)
+ return ret;
+
+ formats = ff_all_samplerates();
+ if (!formats)
+ return AVERROR(ENOMEM);
+ return ff_set_common_samplerates(ctx, formats);
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+ APitchContext *s = ctx->priv;
+
+ av_fft_end(s->fft);
+ s->fft = NULL;
+ av_fft_end(s->ifft);
+ s->ifft = NULL;
+
+ av_frame_free(&s->acc);
+ av_frame_free(&s->magnitude);
+ av_frame_free(&s->phase);
+ av_frame_free(&s->peaks);
+ av_frame_free(&s->new_phase);
+ av_frame_free(&s->last_phase);
+ av_frame_free(&s->osamples);
+ av_frame_free(&s->map);
+ av_frame_free(&s->buffer);
+
+ av_freep(&s->window_func_lut);
+
+ av_audio_fifo_free(s->ififo);
+ s->ififo = NULL;
+
+ swr_free(&s->swr);
+}
+
+static int process_command(AVFilterContext *ctx,
+ const char *cmd,
+ const char *arg,
+ char *res,
+ int res_len,
+ int flags)
+{
+ APitchContext *s = ctx->priv;
+ float tempo;
+ float pitch;
+ int ret = 0;
+
+ if (!strcmp(cmd, "tempo") && av_sscanf(arg, "%f", &tempo) == 1) {
+ s->tempo = av_clipf(tempo, 0.01f, 100.f);
+ ret = set_input_overlap(s, s->tempo / s->pitch);
+ } else if (!strcmp(cmd, "pitch") && av_sscanf(arg, "%f", &pitch) == 1) {
+ pitch = av_clipf(pitch, 0.01f, 100.f);
+ s->pitch_changed = pitch != s->pitch;
+ s->pitch = pitch;
+ ret = set_input_overlap(s, s->tempo / s->pitch);
+ } else {
+ ret = AVERROR(ENOSYS);
+ }
+
+ return ret;
+}
+
+static const AVFilterPad inputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ },
+ { NULL }
+};
+
+static const AVFilterPad outputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .config_props = config_output,
+ },
+ { NULL }
+};
+
+AVFilter ff_af_apitch = {
+ .name = "apitch",
+ .description = NULL_IF_CONFIG_SMALL("Adjust audio pitch and tempo."),
+ .priv_size = sizeof(APitchContext),
+ .priv_class = &apitch_class,
+ .inputs = inputs,
+ .outputs = outputs,
+ .activate = activate,
+ .query_formats = query_formats,
+ .process_command = process_command,
+ .uninit = uninit,
+};
@@ -60,6 +60,7 @@ extern AVFilter ff_af_anull;
extern AVFilter ff_af_apad;
extern AVFilter ff_af_aperms;
extern AVFilter ff_af_aphaser;
+extern AVFilter ff_af_apitch;
extern AVFilter ff_af_apulsator;
extern AVFilter ff_af_arealtime;
extern AVFilter ff_af_aresample;
Signed-off-by: Paul B Mahol <onemda@gmail.com> --- This filter can dynamically change both tempo and pitch of audio. Also scale range is bigger, from 0.01 to 100. --- libavfilter/Makefile | 1 + libavfilter/af_apitch.c | 764 +++++++++++++++++++++++++++++++++++++++ libavfilter/allfilters.c | 1 + 3 files changed, 766 insertions(+) create mode 100644 libavfilter/af_apitch.c